603 lines
21 KiB
C++
603 lines
21 KiB
C++
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/*
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* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "rtp_receiver_audio.h"
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#include <cassert> //assert
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#include <cstring> // memcpy()
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#include <math.h> // pow()
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#include "critical_section_wrapper.h"
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namespace webrtc {
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RTPReceiverAudio::RTPReceiverAudio(const WebRtc_Word32 id):
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_id(id),
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_lastReceivedFrequency(8000),
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_telephoneEvent(false),
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_telephoneEventForwardToDecoder(false),
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_telephoneEventDetectEndOfTone(false),
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_telephoneEventPayloadType(-1),
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_telephoneEventReported(),
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_cngNBPayloadType(-1),
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_cngWBPayloadType(-1),
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_cngSWBPayloadType(-1),
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_cngPayloadType(-1),
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_G722PayloadType(-1),
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_lastReceivedG722(false),
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_criticalSectionFeedback(*CriticalSectionWrapper::CreateCriticalSection()),
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_cbAudioFeedback(NULL)
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{
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}
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RTPReceiverAudio::~RTPReceiverAudio()
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{
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delete &_criticalSectionFeedback;
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}
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WebRtc_Word32
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RTPReceiverAudio::Init()
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{
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_lastReceivedFrequency = 8000;
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_telephoneEvent = false;
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_telephoneEventForwardToDecoder = false;
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_telephoneEventDetectEndOfTone = false;
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_telephoneEventPayloadType = -1;
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while(_telephoneEventReported.Size() > 0)
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{
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_telephoneEventReported.Erase(_telephoneEventReported.First());
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}
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_cngNBPayloadType = -1;
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_cngWBPayloadType = -1;
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_cngSWBPayloadType = -1;
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_cngPayloadType = -1;
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_G722PayloadType = -1;
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_lastReceivedG722 = false;
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return 0;
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}
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void
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RTPReceiverAudio::ChangeUniqueId(const WebRtc_Word32 id)
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{
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_id = id;
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}
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WebRtc_Word32
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RTPReceiverAudio::RegisterIncomingAudioCallback(RtpAudioFeedback* incomingMessagesCallback)
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{
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CriticalSectionScoped lock(_criticalSectionFeedback);
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_cbAudioFeedback = incomingMessagesCallback;
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return 0;
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}
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WebRtc_UWord32
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RTPReceiverAudio::AudioFrequency() const
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{
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if(_lastReceivedG722)
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{
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return 8000;
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}
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return _lastReceivedFrequency;
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}
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// Outband TelephoneEvent(DTMF) detection
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WebRtc_Word32
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RTPReceiverAudio::SetTelephoneEventStatus(const bool enable,
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const bool forwardToDecoder,
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const bool detectEndOfTone)
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{
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_telephoneEvent= enable;
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_telephoneEventDetectEndOfTone = detectEndOfTone;
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_telephoneEventForwardToDecoder = forwardToDecoder;
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return 0;
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}
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// Is outband TelephoneEvent(DTMF) turned on/off?
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bool
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RTPReceiverAudio::TelephoneEvent() const
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{
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return _telephoneEvent;
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}
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// Is forwarding of outband telephone events turned on/off?
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bool
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RTPReceiverAudio::TelephoneEventForwardToDecoder() const
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{
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return _telephoneEventForwardToDecoder;
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}
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bool
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RTPReceiverAudio::TelephoneEventPayloadType(const WebRtc_Word8 payloadType) const
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{
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return (_telephoneEventPayloadType == payloadType)?true:false;
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}
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bool
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RTPReceiverAudio::CNGPayloadType(const WebRtc_Word8 payloadType,
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WebRtc_UWord32& frequency)
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{
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// we can have three CNG on 8000Hz, 16000Hz and 32000Hz
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if(_cngNBPayloadType == payloadType)
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{
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frequency = 8000;
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if ((_cngPayloadType != -1) &&(_cngPayloadType !=_cngNBPayloadType))
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{
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ResetStatistics();
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}
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_cngPayloadType = _cngNBPayloadType;
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return true;
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} else if(_cngWBPayloadType == payloadType)
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{
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// if last received codec is G.722 we must use frequency 8000
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if(_lastReceivedG722)
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{
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frequency = 8000;
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} else
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{
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frequency = 16000;
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}
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if ((_cngPayloadType != -1) &&(_cngPayloadType !=_cngWBPayloadType))
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{
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ResetStatistics();
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}
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_cngPayloadType = _cngWBPayloadType;
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return true;
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}else if(_cngSWBPayloadType == payloadType)
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{
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frequency = 32000;
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if ((_cngPayloadType != -1) &&(_cngPayloadType !=_cngSWBPayloadType))
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{
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ResetStatistics();
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}
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_cngPayloadType = _cngSWBPayloadType;
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return true;
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}else
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{
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// not CNG
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if(_G722PayloadType == payloadType)
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{
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_lastReceivedG722 = true;
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}else
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{
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_lastReceivedG722 = false;
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}
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}
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return false;
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}
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/*
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Sample based or frame based codecs based on RFC 3551
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NOTE! There is one error in the RFC, stating G.722 uses 8 bits/samples.
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The correct rate is 4 bits/sample.
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name of sampling default
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encoding sample/frame bits/sample rate ms/frame ms/packet
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Sample based audio codecs
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DVI4 sample 4 var. 20
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G722 sample 4 16,000 20
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G726-40 sample 5 8,000 20
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G726-32 sample 4 8,000 20
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G726-24 sample 3 8,000 20
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G726-16 sample 2 8,000 20
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L8 sample 8 var. 20
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L16 sample 16 var. 20
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PCMA sample 8 var. 20
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PCMU sample 8 var. 20
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Frame based audio codecs
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G723 frame N/A 8,000 30 30
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G728 frame N/A 8,000 2.5 20
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G729 frame N/A 8,000 10 20
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G729D frame N/A 8,000 10 20
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G729E frame N/A 8,000 10 20
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GSM frame N/A 8,000 20 20
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GSM-EFR frame N/A 8,000 20 20
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LPC frame N/A 8,000 20 20
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MPA frame N/A var. var.
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G7221 frame N/A
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*/
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ModuleRTPUtility::Payload*
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RTPReceiverAudio::RegisterReceiveAudioPayload(const WebRtc_Word8 payloadName[RTP_PAYLOAD_NAME_SIZE],
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const WebRtc_Word8 payloadType,
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const WebRtc_UWord32 frequency,
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const WebRtc_UWord8 channels,
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const WebRtc_UWord32 rate)
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{
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WebRtc_Word32 length = (WebRtc_Word32)strlen(payloadName);
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if(length > RTP_PAYLOAD_NAME_SIZE)
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{
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assert(false);
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return NULL;
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}
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if (ModuleRTPUtility::StringCompare(payloadName,"telephone-event",15))
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{
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_telephoneEventPayloadType = payloadType;
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}
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if (ModuleRTPUtility::StringCompare(payloadName,"cn",2))
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{
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// we can have three CNG on 8000Hz, 16000Hz and 32000Hz
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if(frequency == 8000)
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{
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_cngNBPayloadType = payloadType;
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} else if(frequency == 16000)
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{
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_cngWBPayloadType = payloadType;
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} else if(frequency == 32000)
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{
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_cngSWBPayloadType = payloadType;
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}else
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{
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assert(false);
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return NULL;
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}
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}
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WebRtc_UWord8 bitsPerSample = 0; // zero implies frame based
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if (ModuleRTPUtility::StringCompare(payloadName,"DVI4",4))
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{
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bitsPerSample = 4;
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} else if(ModuleRTPUtility::StringCompare(payloadName,"G722",4))
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{
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if(ModuleRTPUtility::StringCompare(payloadName,"G7221",5))
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{
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// frame based
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} else
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{
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_G722PayloadType = payloadType;
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bitsPerSample = 4;
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}
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} else if(ModuleRTPUtility::StringCompare(payloadName,"G726-40",7))
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{
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bitsPerSample = 5;
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} else if(ModuleRTPUtility::StringCompare(payloadName,"G726-32",7))
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{
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bitsPerSample = 4;
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} else if(ModuleRTPUtility::StringCompare(payloadName,"G726-24",7))
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{
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bitsPerSample = 3;
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} else if(ModuleRTPUtility::StringCompare(payloadName,"G726-16",7))
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{
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bitsPerSample = 2;
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} else if(ModuleRTPUtility::StringCompare(payloadName,"L8",2))
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{
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bitsPerSample = 8;
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} else if(ModuleRTPUtility::StringCompare(payloadName,"L16",3))
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{
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bitsPerSample = 16;
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} else if(ModuleRTPUtility::StringCompare(payloadName,"PCMU",4))
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{
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bitsPerSample = 8;
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} else if(ModuleRTPUtility::StringCompare(payloadName,"PCMA",4))
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{
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bitsPerSample = 8;
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}
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ModuleRTPUtility::Payload* payload = new ModuleRTPUtility::Payload;
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memcpy(payload->name, payloadName, length+1);
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payload->typeSpecific.Audio.frequency = frequency;
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payload->typeSpecific.Audio.channels = channels;
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payload->typeSpecific.Audio.bitsPerSample = bitsPerSample;
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payload->typeSpecific.Audio.rate = rate;
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payload->audio = true;
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return payload;
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}
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// we are not allowed to have any critsects when calling CallbackOfReceivedPayloadData
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WebRtc_Word32
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RTPReceiverAudio::ParseAudioCodecSpecific(WebRtcRTPHeader* rtpHeader,
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const WebRtc_UWord8* payloadData,
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const WebRtc_UWord16 payloadLength,
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const ModuleRTPUtility::AudioPayload& audioSpecific,
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const bool isRED)
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{
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WebRtc_UWord8 newEvents[MAX_NUMBER_OF_PARALLEL_TELEPHONE_EVENTS];
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WebRtc_UWord8 removedEvents[MAX_NUMBER_OF_PARALLEL_TELEPHONE_EVENTS];
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WebRtc_UWord8 numberOfNewEvents = 0;
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WebRtc_UWord8 numberOfRemovedEvents = 0;
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bool telephoneEventPacket = TelephoneEventPayloadType(rtpHeader->header.payloadType);
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if(payloadLength == 0)
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{
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return 0;
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}
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{
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CriticalSectionScoped lock(_criticalSectionFeedback);
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if(telephoneEventPacket)
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{
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// RFC 4733 2.3
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/*
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0 1 2 3
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0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
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+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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| event |E|R| volume | duration |
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+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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*/
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if(payloadLength % 4 != 0)
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{
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return -1;
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}
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WebRtc_UWord8 numberOfEvents = payloadLength / 4;
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// sanity
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if(numberOfEvents >= MAX_NUMBER_OF_PARALLEL_TELEPHONE_EVENTS)
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{
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numberOfEvents = MAX_NUMBER_OF_PARALLEL_TELEPHONE_EVENTS;
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}
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for (int n = 0; n < numberOfEvents; n++)
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{
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bool end = (payloadData[(4*n)+1] & 0x80)? true:false;
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if(_telephoneEventReported.Find(payloadData[4*n]) != NULL)
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{
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// we have already seen this event
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if(end)
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{
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removedEvents[numberOfRemovedEvents]= payloadData[4*n];
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numberOfRemovedEvents++;
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_telephoneEventReported.Erase(payloadData[4*n]);
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}
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}else
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{
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if(end)
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{
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// don't add if it's a end of a tone
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}else
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{
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newEvents[numberOfNewEvents] = payloadData[4*n];
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numberOfNewEvents++;
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_telephoneEventReported.Insert(payloadData[4*n],NULL);
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}
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}
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}
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// RFC 4733 2.5.1.3 & 2.5.2.3 Long-Duration Events
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// should not be a problem since we don't care about the duration
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// RFC 4733 See 2.5.1.5. & 2.5.2.4. Multiple Events in a Packet
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}
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if(_telephoneEvent && _cbAudioFeedback)
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{
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for (int n = 0; n < numberOfNewEvents; n++)
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{
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_cbAudioFeedback->OnReceivedTelephoneEvent(_id, newEvents[n], false);
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}
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if(_telephoneEventDetectEndOfTone)
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{
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for (int n = 0; n < numberOfRemovedEvents; n++)
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{
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_cbAudioFeedback->OnReceivedTelephoneEvent(_id, removedEvents[n], true);
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}
|
||
|
|
}
|
||
|
|
}
|
||
|
|
}
|
||
|
|
if(! telephoneEventPacket )
|
||
|
|
{
|
||
|
|
_lastReceivedFrequency = audioSpecific.frequency;
|
||
|
|
}
|
||
|
|
|
||
|
|
// Check if this is a CNG packet, receiver might want to know
|
||
|
|
WebRtc_UWord32 dummy;
|
||
|
|
if(CNGPayloadType(rtpHeader->header.payloadType, dummy))
|
||
|
|
{
|
||
|
|
rtpHeader->type.Audio.isCNG=true;
|
||
|
|
rtpHeader->frameType = kAudioFrameCN;
|
||
|
|
}else
|
||
|
|
{
|
||
|
|
rtpHeader->frameType = kAudioFrameSpeech;
|
||
|
|
rtpHeader->type.Audio.isCNG=false;
|
||
|
|
}
|
||
|
|
|
||
|
|
// check if it's a DTMF event, hence something we can playout
|
||
|
|
if(telephoneEventPacket)
|
||
|
|
{
|
||
|
|
if(!_telephoneEventForwardToDecoder)
|
||
|
|
{
|
||
|
|
// don't forward event to decoder
|
||
|
|
return 0;
|
||
|
|
}
|
||
|
|
MapItem* first = _telephoneEventReported.First();
|
||
|
|
if(first && first->GetId() > 15)
|
||
|
|
{
|
||
|
|
// don't forward non DTMF events
|
||
|
|
return 0;
|
||
|
|
}
|
||
|
|
}
|
||
|
|
if(isRED && !(payloadData[0] & 0x80))
|
||
|
|
{
|
||
|
|
// we recive only one frame packed in a RED packet remove the RED wrapper
|
||
|
|
rtpHeader->header.payloadType = payloadData[0];
|
||
|
|
|
||
|
|
// only one frame in the RED strip the one byte to help NetEq
|
||
|
|
return CallbackOfReceivedPayloadData(payloadData+1,
|
||
|
|
payloadLength-1,
|
||
|
|
rtpHeader);
|
||
|
|
}
|
||
|
|
if(audioSpecific.channels > 1)
|
||
|
|
{
|
||
|
|
WebRtc_Word32 retVal = 0;
|
||
|
|
WebRtc_UWord16 channelLength = payloadLength/audioSpecific.channels;
|
||
|
|
|
||
|
|
if(audioSpecific.bitsPerSample > 0)
|
||
|
|
{
|
||
|
|
// sanity
|
||
|
|
assert((payloadLength*8)%audioSpecific.bitsPerSample == 0);
|
||
|
|
|
||
|
|
// sample based codec
|
||
|
|
|
||
|
|
// build matrix
|
||
|
|
WebRtc_UWord8 matrix[IP_PACKET_SIZE];
|
||
|
|
WebRtc_UWord32 offsetBytes = 0;
|
||
|
|
WebRtc_UWord32 offsetBytesInsert = 0;
|
||
|
|
// initialize matrix to 0
|
||
|
|
memset(matrix, 0, audioSpecific.channels*channelLength);
|
||
|
|
|
||
|
|
switch(audioSpecific.bitsPerSample)
|
||
|
|
{
|
||
|
|
case 1:
|
||
|
|
case 2:
|
||
|
|
case 3:
|
||
|
|
case 4:
|
||
|
|
case 5:
|
||
|
|
case 6:
|
||
|
|
case 7:
|
||
|
|
{
|
||
|
|
WebRtc_UWord32 offsetSamples = 0;
|
||
|
|
WebRtc_UWord32 offsetSamplesInsert = 0;
|
||
|
|
WebRtc_UWord16 bitMask = (WebRtc_UWord16)ModuleRTPUtility::pow2(audioSpecific.bitsPerSample)-1;
|
||
|
|
WebRtc_UWord16 samplesPerChannel =payloadLength*8/audioSpecific.bitsPerSample/audioSpecific.channels;
|
||
|
|
|
||
|
|
for(WebRtc_UWord32 i = 0; i < samplesPerChannel; i++)
|
||
|
|
{
|
||
|
|
WebRtc_UWord8 insertShift = (WebRtc_UWord8)((offsetSamplesInsert+audioSpecific.bitsPerSample)%16);
|
||
|
|
insertShift = 16 - insertShift; // inverse the calculation
|
||
|
|
|
||
|
|
for(WebRtc_UWord32 j = 0; j < audioSpecific.channels; j++)
|
||
|
|
{
|
||
|
|
// get sample
|
||
|
|
WebRtc_UWord16 s = payloadData[offsetBytes] << 8;
|
||
|
|
|
||
|
|
// check that we don't read outside the memory
|
||
|
|
if(offsetBytes < (WebRtc_UWord32)payloadLength -2)
|
||
|
|
{
|
||
|
|
s += payloadData[offsetBytes+1];
|
||
|
|
}
|
||
|
|
|
||
|
|
WebRtc_UWord8 readShift = (WebRtc_UWord8)((offsetSamples+audioSpecific.bitsPerSample)%16);
|
||
|
|
readShift = 16 - readShift; // inverse the calculation
|
||
|
|
s >>= readShift;
|
||
|
|
s &= bitMask;
|
||
|
|
|
||
|
|
// prepare for reading next sample
|
||
|
|
offsetSamples += audioSpecific.bitsPerSample;
|
||
|
|
if(readShift <= audioSpecific.bitsPerSample)
|
||
|
|
{
|
||
|
|
// next does not fitt
|
||
|
|
// or fitt exactly
|
||
|
|
offsetSamples -= 8;
|
||
|
|
offsetBytes++;
|
||
|
|
}
|
||
|
|
|
||
|
|
// insert sample into matrix
|
||
|
|
WebRtc_UWord32 columOffset = j*channelLength;
|
||
|
|
|
||
|
|
WebRtc_UWord16 insert = s << insertShift;
|
||
|
|
#if defined(WEBRTC_LITTLE_ENDIAN)
|
||
|
|
matrix[columOffset+offsetBytesInsert] |= static_cast<WebRtc_UWord8>(insert>>8);
|
||
|
|
matrix[columOffset+offsetBytesInsert+1] |= static_cast<WebRtc_UWord8>(insert);
|
||
|
|
#else
|
||
|
|
WebRtc_UWord16* matrixU16 = (WebRtc_UWord16*)&(matrix[columOffset+offsetBytesInsert]);
|
||
|
|
matrixU16[0] |= (s << insertShift);
|
||
|
|
#endif
|
||
|
|
}
|
||
|
|
// prepare for writing next sample
|
||
|
|
offsetSamplesInsert += audioSpecific.bitsPerSample;
|
||
|
|
if(insertShift <= audioSpecific.bitsPerSample)
|
||
|
|
{
|
||
|
|
// next does not fitt
|
||
|
|
// or fitt exactly
|
||
|
|
offsetSamplesInsert -= 8;
|
||
|
|
offsetBytesInsert++;
|
||
|
|
}
|
||
|
|
}
|
||
|
|
}
|
||
|
|
break;
|
||
|
|
case 8:
|
||
|
|
{
|
||
|
|
WebRtc_UWord32 sample = 0;
|
||
|
|
for(WebRtc_UWord32 i = 0; i < channelLength; i++)
|
||
|
|
{
|
||
|
|
for(WebRtc_UWord32 j = 0; j < audioSpecific.channels; j++)
|
||
|
|
{
|
||
|
|
WebRtc_UWord32 columOffset = j*channelLength;
|
||
|
|
matrix[columOffset + i] = payloadData[sample++];
|
||
|
|
}
|
||
|
|
}
|
||
|
|
}
|
||
|
|
break;
|
||
|
|
case 16:
|
||
|
|
{
|
||
|
|
WebRtc_UWord32 sample = 0;
|
||
|
|
for(WebRtc_UWord32 i = 0; i < channelLength; i +=2)
|
||
|
|
{
|
||
|
|
for(WebRtc_UWord32 j = 0; j < audioSpecific.channels; j++)
|
||
|
|
{
|
||
|
|
WebRtc_UWord32 columOffset = j*channelLength;
|
||
|
|
matrix[columOffset + i] = payloadData[sample++];
|
||
|
|
matrix[columOffset + i + 1] = payloadData[sample++];
|
||
|
|
}
|
||
|
|
}
|
||
|
|
}
|
||
|
|
break;
|
||
|
|
default:
|
||
|
|
assert(false);
|
||
|
|
return -1;
|
||
|
|
}
|
||
|
|
// we support 16 bits sample
|
||
|
|
// callback for all channels
|
||
|
|
for(int channel = 0; channel < audioSpecific.channels && retVal == 0; channel++)
|
||
|
|
{
|
||
|
|
// one callback per channel
|
||
|
|
rtpHeader->type.Audio.channel = channel+1;
|
||
|
|
|
||
|
|
if(channel == 0)
|
||
|
|
{
|
||
|
|
// include the original packet only in the first callback
|
||
|
|
retVal = CallbackOfReceivedPayloadData(&matrix[channel*channelLength],
|
||
|
|
channelLength,
|
||
|
|
rtpHeader);
|
||
|
|
} else
|
||
|
|
{
|
||
|
|
retVal = CallbackOfReceivedPayloadData(&matrix[channel*channelLength],
|
||
|
|
channelLength,
|
||
|
|
rtpHeader);
|
||
|
|
}
|
||
|
|
}
|
||
|
|
} else
|
||
|
|
{
|
||
|
|
for(int channel = 1; channel <= audioSpecific.channels && retVal == 0; channel++)
|
||
|
|
{
|
||
|
|
// one callback per channel
|
||
|
|
rtpHeader->type.Audio.channel = channel;
|
||
|
|
|
||
|
|
if(channel == 1)
|
||
|
|
{
|
||
|
|
// include the original packet only in the first callback
|
||
|
|
retVal = CallbackOfReceivedPayloadData(payloadData,
|
||
|
|
channelLength,
|
||
|
|
rtpHeader);
|
||
|
|
} else
|
||
|
|
{
|
||
|
|
retVal = CallbackOfReceivedPayloadData(payloadData,
|
||
|
|
channelLength,
|
||
|
|
rtpHeader);
|
||
|
|
}
|
||
|
|
payloadData += channelLength;
|
||
|
|
}
|
||
|
|
}
|
||
|
|
return retVal;
|
||
|
|
}else
|
||
|
|
{
|
||
|
|
rtpHeader->type.Audio.channel = 1;
|
||
|
|
return CallbackOfReceivedPayloadData(payloadData,
|
||
|
|
payloadLength,
|
||
|
|
rtpHeader);
|
||
|
|
}
|
||
|
|
}
|
||
|
|
} // namespace webrtc
|